The present invention relates to a signal processing apparatus and, more particularly to a signal processing apparatus which reduces operation amount in decoding when reproducing a plurality of signals of different sampling frequencies.
According to a DVD audio standard for a DVD-based audio disc, its storage capacity is 4.7 GB. DVD audio recording scheme is PCM, like CD or DVD-ROM and, as for its specification, a sampling frequency indicative of fidelity to original sound with which audio is recorded is 192 KHz at maximum, which is about 4.3 times as high as that (44.1 KHz) of the CD. This enables to record audio of highest quality.
FIG. 3 is a diagram showing conception of a sound field formed by reproduced DVD audio. In the figure, reference numeral 40 denotes a center speaker, and 41a and 41b denote left and right speakers placed at the left of the center speaker 40 and at the right of the same, respectively. Reference numerals 42a and 42b denote left and right surround speakers placed behind an auditor, for increasing realism, and 43 denotes a speaker called xe2x80x9csub-wooferxe2x80x9d, for outputting relatively low sound. According to the DVD audio standard, reproduction can be performed by using 6 speakers (6 channels) even when sampling frequencies and the numbers of quantization bits of respective channels differ from each other. For instance, in configuration shown in FIG. 3, the center speaker 40 and the left and right channel speakers 41a and 41b for which relatively high sound quality is demanded, perform reproduction at 96 KHz, while the left and right surround speakers 42a and 42b and the sub-woofer 43 for which relatively high sound quality is not demanded, perform reproduction at 48 KHz.
By the way, when data of respective channels are to be recorded at 96 KHz and in 24 bits for data of 6 channels, a standard for a maximum transfer rate would be exceeded. Accordingly, it becomes necessary to compress data when recorded. A compression method includes irreversible compression using a psychoacoustic model for use in MPEG or AC3, and xe2x80x9cLossless compressionxe2x80x9d which is capable of completely restoring data to the state before compression by employing entropy coding as reversible compression, such as Huffman coding. In order to reproduce audio of high quality with fidelity to original sound as described above, the Lossless compression is desirably employed. This enables to reproduce audio of high quality of 6 channels at 96 KHz and in 24 bits, in data transfer of the DVD audio. On the other hand, even when the standard for the maximum transfer rate is not exceeded, the Lossless compression enables to record data of 4.7 GB for a long time period.
FIG. 4 is a block diagram showing a conventional DVD audio recording apparatus. For the sake of simplicity, 3 channels are illustrated, although 6 channels are actually used. In FIG. 4, reference numerals 51a and 51b denote upsampling means which receive signals of the channels 2 and 3 at sampling frequencies of 48 KHz, and adapt their respective sampling frequencies to 96 KHz for the channel 1. Reference numeral 50 denotes a timing delay unit for delaying the signal of the channel 1 while the signals of the channel 2 and 3 are upsampled, and 52 denotes a filter circuit for filtering the upsampled signals of the channels 2 and 3 and performing interpolation for them so that they are smoothed. Reference numeral 54 denotes Lossless compression means for performing reversible compression of the signals of the channels 2 and 3 which passed through the filter circuit 52 and the signal of the channel 1 delayed by the delay unit 50. Reference numeral 53 denotes format transformation means for transforming a Lossless-compressed signal into data having a predetermined format which can be written to a recording medium 56, and 55 denotes recording means for recording the compressed data into the recording medium 56.
To upsample the sampling frequencies of the signals of the channels 2 and 3 from 48 KHz to 96 KHz, respectively, with the above-mentioned construction, the upsampling means 51a and 51b insert a predetermined number of xe2x80x9czerosxe2x80x9d into data so that the sampling frequencies are twice higher (48xc3x972=96), and then a filter circuit 52 having a given factor in a subsequent stage replaces the inserted xe2x80x9czeroxe2x80x9d samples with :samples used for smooth interpolation. To be specific, the upsampling process for inserting the samples having xe2x80x9czeroxe2x80x9d values into the data of the signals of the channels 2 and 3 is performed so that the sample having the xe2x80x9czeroxe2x80x9d value is placed in every other sample of the data. While the signals of the channels 2 and 3 are upsampled, the signal of the channel 1 is delayed by the delay unit 50. Instead of the above xe2x80x9c0xe2x80x9d insertion, processing performed by the upsampling means 51a and 51b may be a sample holding process which holds a predetermined number of previous sample data or an interpolation process using straight lines rather than xe2x80x9czerosxe2x80x9d. Here, in the sample holding process, the data of the signals of channels 2 and 3 are interpolated so that after each of the samples constituting that data, a sample having the same value as that sample is placed. For the filter circuit 52, a low pass filter can be realized by a filter such as an FIR (Filter Impulse Response) or an IIR (Infinite Impulse Response). The filter circuit 52 filters the signals output from the upsampling means 51a and 51b by using the above filter.
The outputs of the filter circuit 52 and the output of the delay unit 50 are processed by the Lossless compression means 54 and then processed by the format transformation means 53, and the resulting data is written to the recording medium (DVD audio disc) 56 by using the recording means 55.
To read so created data from the recording medium 56 to reproduce audio, a reproducing apparatus shown in FIG. 5 is used. In FIG. 5, reference numeral 60 denotes reading means for reading data from the recording medium 56, and reference numeral 62 denotes format inverse-transformation means for transforming the read data (Lossless-compressed) into a signal (Lossless-compressed) having a format of reproducible audio signal. Reference numeral 61 denotes compressed-data decompression means for decompressing the data (Lossless-compressed) which has been subjected to the format inverse-transformation, and reference numeral 63 denotes a filter circuit for downsampling predetermined decompressed data as required.
To reproduce the predetermined data decompressed by the compressed-data decompression means 61 at a sampling frequency of 48 KHz downsampled from 98 KHz, with the above-described construction, the filter circuit 63 downsamples this data.
In the conventional signal processing apparatus so constructed, the filter circuit temporarily equalizes the sampling frequencies of the plurality of signals at a recording time, to be recorded in the recording medium, while the filter circuit at a reproducing end changes the sampling frequencies of the predetermined channels into the predetermined sampling frequencies, to output the signals.
In this case, when high precision is required for the filter circuits used in the above processing, the amount of operation therein is noticeably increased, and burden on hardware is correspondingly increased. In addition, the processed signals are reproduced unsatisfactorily.
The present invention is directed to solving the above problem, and an object of the present invention is to provide a signal processing apparatus which is capable of reducing operation amount in filter circuits when processing a plurality of signals of different sampling frequencies, and reproducing all the signals completely.
According to aspect 1 of the present invention, there is provided a signal processing apparatus for encoding a plurality of channel signals of different sampling frequencies to be recorded in a recording medium, comprising: upsampling means for transforming a sampling frequency of a channel signal having a small sampling frequency among the plurality of channel signals of different sampling frequencies into a sampling frequency of a channel signal having a large sampling frequency; a half band filter that receives the channel signal upsampled by the upsampling means as an input; and format transformation means for transforming the channel signal processed by the half band filter and the channel signal having the large sampling frequency into signals having predetermined formats.
According to aspect 2 of the present invention, the signal processing apparatus of aspect 1, further comprises: data compression means for performing Lossless compression to the signal processed by the half band filter and the channel signal having the large sampling frequency.
According to aspect 3 of the present invention, the signal processing apparatus of aspect 1, further comprises: upsampling information description means for describing upsampling information indicating that a sample of the channel signals processed by the half band filter is a sample before the upsampling, into a predetermined area of the recording medium.
According to aspect 4 of the present invention, there is provided a signal processing apparatus for performing decoding to reproduce data recorded in a recording medium, said data being recorded in the recording medium after a sampling frequency of at least one channel signal among a plurality of channel signals of different sampling frequencies is upsampled, and the resulting channel signal is processed by a half band filter and then transformed into a signal having a predetermined format, comprising: data reading means for reading data from the recording medium; format inverse-transformation means for inversely transforming the data read from the data reading means into a signal having a predetermined format; downsampling means for downsampling a sampling frequency of a specified signal among the format inverse-transformed signals; and downsampling control means for instructing the downsampling means to intermittently read the specified signal among the format inverse-transformed signals.
According to aspect 5 of the present invention, in the signal processing apparatus of aspect 4, the signal read by the data reading means is completely reproducible data which has been subjected to Lossless compression, and the apparatus further comprises: data decompression means for decompressing compressed data into data before compression after it is subjected to format inverse-transformation.
According to aspect 6 of the present invention, the signal processing apparatus of aspect 4, further comprises: upsampling information extraction means for detecting upsampling information indicating that data is a sample before upsampling performed by the upsampling means when detecting the upsampled signal data; and downsampling control means for instructing the downsampling means to intermittently read the data according to the information.
According to aspect 7 of the present invention, in the signal processing apparatus of aspect 6, the upsampling information is recorded in a predetermined area of the recording medium as an information indicating data is a sample before upsampling performed by the upsampling means, and the downsampling means reads one of odd-numbered data and even-numbered data according to the upsampling information, thereby to perform said intermittent reading.
As described above, according to aspect 1 of the present invention, there is provided a signal processing apparatus for encoding a plurality of channel signals of different sampling frequencies to be recorded in a recording medium, comprising: upsampling means for transforming a sampling frequency of a channel signal having a small sampling frequency among the plurality of channel signals of different sampling frequencies into a sampling frequency of a channel signal having a large sampling frequency; a half band filter that receives the channel signal upsampled by the upsampling means as an input; and format transformation means for transforming the channel signal processed by the half band filter and the channel signal having the large sampling frequency into signals having predetermined formats. Therefore, the operation amount of the filter in upsampling can be reduced about by half.
According to aspect 2 of the present invention, the signal processing apparatus of aspect 1, further comprises: data compression means for performing Lossless compression to the signal processed by the half band filter and the channel signal having the large sampling frequency. Therefore, much data can be recorded in the recording medium without degrading sound quality.
According to aspect 3 of the present invention, the signal processing apparatus of aspect 1, further comprises: upsampling information description means for describing upsampling information indicating that a sample of the channel signal processed by the half band filter is a sample before the upsampling, into a predetermined area of the recording medium. Therefore, when reproducing a signal from the recording medium later, an original shape of the upsampled signal can be recognized, and data can be read intermittently with reliability.
According to aspect 4 of the present invention, there is provided a signal processing apparatus for performing decoding to reproduce data recorded in a recording medium, said data being recorded in the recording medium after a sampling frequency of at least one channel signal among a plurality of channel signals of different sampling frequencies is upsampled, and the resulting channel signal is processed by a half band filter and then transformed into a signal having a predetermined format, comprising: data reading means for reading data from the recording medium; format inverse-transformation means for inversely transforming the data read from the data reading means into a signal having a predetermined format; downsampling means for downsampling a sampling frequency of a specified signal among the format inverse-transformed signals; and downsampling control means for instructing the downsampling means to intermittently read the specified signal among the format inverse-transform ed signals. Therefore, downsampling can be implemented by only performing intermittent reading, and thereby a filter at a reproducing end is dispensed with, which significantly reduces operation amount.
According to aspect 5 of the present invention, in the signal processing apparatus of aspect 4, the signal read by the data reading means is completely reproducible data which has been subjected to Lossless compression, and the apparatus further comprises: data decompression means for decompressing compressed data into data before compression after it is subjected to format inverse-transformation. Therefore, when reproducing the compressed data, its sound quality is not degraded at all.
According to aspect 6 of the present invention, the signal processing apparatus of aspect 4, further comprises: upsampling information extraction means for detecting upsampling information indicating that data is a sample before upsampling performed by the upsampling means when detecting the upsampled signal data; and downsampling control means for instructing the downsampling means to intermittently read the data according to the information. Therefore, an original shape of the upsampled signal can be recognized, and data can be read intermittently with reliability.
According to aspect 7 of the present invention, in the signal processing apparatus of aspect 6, the upsampling information is recorded in a predetermined area of the recording medium as an information indicating data is a sample before upsampling performed by the upsampling means, and the downsampling means reads one of odd-numbered data and even-numbered data according to the upsampling information, thereby to perform said intermittent reading. Therefore, the signal can be easily downsampled with a simple circuit.